Archive for the ‘studio’ Category

i keep changing my mind

I was going to post about the DEVO/Blondie show I saw a couple of months ago because I want to talk about nostalgia acts vs. acts which may have been around a while but are still going, and I bet you can guess which is which here, but I’m just not feelin’ it. So NONE FOR YOU! Today anyway. Later.

Plus I’m crazy busy because today is laundry day and cleaning day and also I built a new remote-controled ON AIR/OFF AIR sign for downstairs last night that I modified a little this morning (I changed the plug so it’s out of the way):


 

…AND I have someone coming in at 2pm for a vocals recording session, so I just don’t have time.

I’ll talk about the sign in a DIY post on Wednesday or maybe the Wednesday after. It’s REMOTE CONTROLLED! And very, very 20th Century, to echo a comment made on Livejournal about my Big Board and recording organisation system. Yes, yes, it is.

Somebody buy me a free tablet I can just hang on the wall instead and I’ll show you the modern solution that I’ve already built. But that one needs a smarter display, something that can stay powered for hours and listen to a webserver and and and.

More Wednesday. ^_^

got some guitar to record today

Got some guitar to record today – not me, I don’t play that, unless you count bass, which you shouldn’t – so have some fun things.

If you haven’t seen it yet, watch this Air New Zealand air safety video. YES, REALLY. I know. Trust me here.

I posted a video a little while ago about studio treatment, getting the room all nice and under control for recording. It was a DIY thing. I also made this 360° view of my home studio, but didn’t embed it. That’s from room centre. As you can see, it’s a small space! But it works.

STERN MICROPHONE IS VERY STERN. Also very Russian:


Во славу Родины, броситься под немецкие танки!

Finally, since we’re recording guitar today, here’s a 3D printed guitar. Change that to an Irish Bouzouki, take it on the road, you pretty much have everything I ever post about here. XD

Soundtrack album work all weekend for me – I’m going to be editing together some learning tracks for our musicians. What’ve you got?

fine-tuning the room

I’ve been fine-tuning the studio in preparation for work on the soundtrack, plus I have some other work coming up for Leannan Sidhe – their Roses and Ruin project and some second-studio work on their next full studio album. (Having their main engineer and studio be in Oregon when they’re in Seattle is problematic from a scheduling standpoint, so I’m helping out.)

And I have to tell you something: before the latest round of adjustments, I would never have put the words “Bose” and “precise” in the same sentence together. Not without also including the word “not,” anyway. I even dragged Anna upstairs and into the room to listen and she was all, “…<blink> wow.”

Remember my Bose? I’ve talked about them before. They’re old 301s, from this post where I talk about how terrible they sounded in the living room because of the room’s odd shape – see the link, it’s relevant. They sounded much better in the studio – a rectangular room – but still not the way you’d think they should.

Turns out, the way to get their best performance is to put them in a finely-tuned recording studio. IS THAT ALL? WHO KNEW!? ノಠ益ಠ)ノ彡┻━┻

But I put on a test track and was suddenly transported back to my radio days, because they finally sounded like pro gear. I was… not expecting that! It’s not even that they’re just high-end precise; they’re unexpectedly crisp in the low end, too. You particularly hear it in percussion and bass guitar; suddenly, I’m hearing things I’ve never heard before, which means they’re worth having as monitors. For special cases.

Anyway, here’s a video showing the current state of studio tuning. It’s short, and annotated heavily. Enjoy!

Oh, and since people have asked, I will indeed talk about the Big Board, probably next week. Or the week after, I have another topic queued up also. Advance reading (or spoiler, if you prefer): “Heijunka box.”

 


This post is part of The DIY Studio Buildout Series, on building out a home recording studio.

studio buildout part 7: Jeff Bohnhoff on room conditioning

Hey, DIYers! Today we have something special for you.

This series has always been about sharing information and people doing things. It’s part of the punk aesthetic, it’s part of particpatory culture, it’s part of maker culture, it’s part of the filk aesthetic – and a part I really like.

So I’m really pleased to feature this post by Jeff Bohnhoff, who will be writing about room conditioning. I’ll let Tony Fabris of the band Vixy & Tony introduce him for you, since he does such a good job of it:

Jeff and his wife Maya have been in rock and folk music all their lives, and have been producing record albums for more than 30 years, so they know a thing or two about both home studio and pro studio recording. Jeff works at Apple as part of the support team for their flagship pro audio recording product, “Logic,” so he knows a thing or two about audio recording software.

He’s produced dozens of amazing-sounding albums, both for himself and for others, so he knows how to get good sound. He also happens to be a brilliant parodist, and Jeff & Maya’s parody albums are characterized by hyper-accurate reconstructions of classic pop songs. He also taught me how to play guitar.

Now, without further ado: Jeff Bohnhoff.


The Sound of Your Studio
by Jeff Bohnhoff, Mystic Fig Studios

So, the topic du jour is acoustics. Specifically, the sound of your studio.

We recording engineers have a natural tendency to geek out over gear – whether we purchase it, make it, or improvise it from inexpensive bits and pieces. Gear is fun, and it’s an important part of making good recordings, but it’s only part of the picture, and not even the most important part.

In fact, better mics, preamps, etc. may lead to worse sounding recordings. “What?!” you say, “how can that possibly be? That Jeff, he’s CRAZY!” Well, okay, guilty as charged, but I stand by the statement. Here’s my reasoning:

When you put a microphone into a room and record something, the acoustic signature of the room is like a fingerprint that covers everything you record. It cannot be removed with any amount of EQ or processing – believe me, I’ve tried. Sound recorded in a bad room is like a white napkin handled by your mechanic after he’s swapped out the oil pan on a ’62 Rambler. The fingerprints are greasy, and will not wash out. Period.

The sad truth is, that your room probably sounds bad. Probably really bad. Most bedrooms, garages, dens, etc simply were not designed to be acoustically pleasing. With the availability of relatively inexpensive, good quality recording gear, this is the main difference between most home studios and commercial facilities. Good facilities have rooms that are designed from the ground up to sound good; everything from the dimensions of the room, the construction methods, to the materials and treatments on the floor, walls and ceiling are designed to eliminate room resonances, slap-back, bass build up, and so on.

So, assuming you are recording in your spare bedroom, why does using better gear – especially a better mic – often lead to worse sounding recordings? A better (i.e. more sensitive) mic “hears” more detail, and picks up more spatial information from the room it’s in. It’s typically more accurate, and does a better job of revealing everything about the source you’re recording and the room it’s in, flaws and all – sort of like the way your HD TV lets you see every pore and blemish on the face of your favorite reality star on Lifestyles of the Vacuous and Incredibly Boring.

Okay – we’ve established the problem, now what’s the solution? Unfortunately, there is no easy, complete solution or panacea. However, there are some practical steps you can take to improve things. First some ideas that involve no modification to the room:

  1. Understand the pick up patterns of your microphones and how positioning affects the sound.

    Most mics have a cardioid pattern. This means they are most sensitive directly in front of the mic, with lobes that extend part way around the back, and almost no pick up directly behind the mic. The shape somewhat resembles a heart, which is where the “cardio” in the word cardioid comes from.


    Cardioid pick up pattern. You’ll see this on many microphone spec sheets.

    The advantage of this pick up pattern is that it hears less of the room than most other patterns, and so can be more useful in challenging acoustical environments. The downside is that cardioid mics exhibit “proximity effect,” which a bass and low mid frequency emphasis when the source is closer to the mic.

    This can be a problem because many small rooms sound very congested and bassy to start with, so even though getting closer to the mic means the source ls louder relative to the room sound (taking the room more out of the sonic picture), the closer you get to the mic, the more low and low-mid cruft you have to deal with.

    Hypercardioid mics (not terribly common) pick up even less sound from the side (but slightly more directly from the rear), but exhibit an even stronger proximity effect than cardioid mics. Omnidirectional mics, as the name implies, pick up equally well in all directions. This means they “hear” the room very well. On the plus side, they have no proximity effect at all. Mics with a figure-8 pick up pattern hear from the from and back equally well, or close to it, with little or no pick up on either side. In my opinion, these are the hardest to use in a challenging room. They “hear” a lot of the room, and they exhibit a fairly strong proximity effect.

    The key here is to experiment with your mics, and find the best pattern and distance from the mic to minimize the problems with your room. This will vary from room to room, mic to mic and even song to song. You may find that singing a foot or more away from a cardioid mic works best for you, or perhaps singing a few inches from an omni mic will work better in other cases. Just remember that the closer you are to the mic, the more source and less room will be recorded. Likewise, the narrower the pick up pattern, the more source and less room, but getting too close may result in too much low mid and bass. Experiment!

  2. Understand your room.

    Rooms do not sound uniform at all positions. If you set your mics up, and just can’t get a sound you like, try other positions in the room. You just may find a location that sounds good, or at least better. If that means contorting yourself into the corner by the bookshelf while standing on one foot, well this is for art, buddy, so suck it up!

  3. Isolate yourself from the room.

    Get some heavy quilts or drapes and some mic stands, and make a tent. Set your mics up inside the tent. This will certainly not give a very lively sound, but it may be better than the sound of the room, and you can have a friend over and tell scary stories in your tent. My favorite is the one about the recording engineer with a hook for a hand, but I digress…

If you can’t tame your room with mic choice or positioning, then you may want to treat it, to make it sound better. This is a complex issue, and before I say anything more, I feel compelled to offer a huge disclaimer.

I am not an expert on this subject, and may well be full of crap. Debates on how to properly treat a room rage in many corners of the internet, among real, trained acoustic engineers, so-called experts, and people who have no idea what they’re talking about, but have access to a working keyboard and an internet connection. I make no pretense at being an acoustician, so I will not be offering pat solutions and miracle cures. Quite honestly, just raising awareness that this is an issue at all is my main goal here.

With all those caveats in mind, I would like to mention some of the issues that many rooms exhibit, and will offer some general ideas on how a home recording engineer might deal with them, and links to some resources for further, more authoritative, information. I’ll be looking at two basic strategies for dealing with acoustic problems: absorption and diffusion.

The principle of absorption is that you mechanically trap sound waves, usually with some sort of material that causes friction, and converts the energy of the sound wave into heat. Absorption is great if the room is just too live. You usually want to avoid overdoing it though, or your room will sound as dead as the annual Christmas party at the office of Q.R. Fishwell, CPA.


Bass absorption panels for Mystic Fig Studios, unmounted

Diffusion, on the other hand, works by breaking up the waveform and dispersing it non-linearly back into the room. This can deal with phase issues, room modes and comb filtering, without overly deadening the room. The downside is that diffusors are relatively expensive to buy, and are generally harder to build than absorbers.

As with absorbers, the lower the effective frequency you want to deal with with, the larger the diffusor needs to be. One of the common types you will see is a “skyline” diffusor. These use “wells” of various heights (based on prime number sequences) to break up waveforms. They are also known as “primitive root diffusors.”

This page has a calculator that lets you plug in a frequency range you want to control, and a number of columns and rows of wells, which are 2″ x 2″ wood, cut to various heights.

The calculator gives you a grid that shows what height each 2″ x 2″ block (in reality 1.5″ x 1.5″, because lumber and math apparently had a disagreement at some point in the past) should be in each position to diffuse the range of frequencies you entered into the formula.

This page has some images of diffusers built using this method. These things don’t take a huge amount of skill to build, but it’s a bit tedious and time consuming.

One possibly easy to get source of diffusion is bookshelves. (But they have to be full of books – you do read, don’t you?) It works especially well if the depths of the books are different, sort of a faux skyline diffusor. Place the shelves opposite the sound source for most effect.


Skyline diffusor

Let’s look at some of the problems a typical room might exhibit:

Bass buildup: Bass frequencies tend to collect in corners, and where the walls meet the ceiling and floor. It’s especially problematic in smaller rooms, because you just don’t have the space to get the mics or speakers all that far away from the corners and edges where the bass is being accentuated.

This problem can be treated with bass traps, which are treatments designed to absorb low frequencies. Here’s the thing: when it comes to bass, physics is working against us. Because bass frequencies are long, it takes a lot of thickness to properly absorb them. Most of the foam based treatments you can buy are pretty much useless below 250 Hz or so.

However, it’s not that hard to build some good bass traps yourself. I built a bunch for my studio using instructions from a video I found on Youtube. These traps use Corning 703 rigid fiberglass insulation as the basic material. You won’t find it at your local mega-lumber yard / home store. The best place is from a commercial insulation supplier. Chances are you won’t be the first studio or home theater DIY’er to darken their door, and they’ll be happy to help you.

Rigid fiberglass comes in 24″ by 48″ by 2″ sheets. Each trap uses two sheets, for a thickness of 4″. The fiberglass is laid over a frame built of 1″x 2″ pine that’s covered with a sheet of cheap fabric. You then cover the fiberglass with some nice fabric (something that tastefully matches your studio decor). You end up with something that sort of looks like a very firm mattress. I mounted mine straddling the joint where the ceiling meets the wall behind my monitors and straddling the joints from floor to ceiling in the room’s corners.


Bass traps at Mystic Fig Studios (click to enlarge)

Flutter echoes: These are “sproingy” metallic sounding echoes, especially noticeable with sounds that have sharp transients, like hand claps. These are generally caused by reflections between the floor and ceiling. The cure is to put some absorbers and/or diffusors on the ceiling.

One approach to absorbers would be to build something like the bass traps, but using Corning 701, which is less dense than the 703, and perhaps only using one 2″ sheet. Unless the flutter is really bad, you probably would aim to cover only 40-50% of the ceiling, in a checkerboard pattern. (I know they’re rectangles, but you get the picture.)

If you want to buy something, then the commercial studio foam from the usual suspects is pretty good for dealing with flutter echoes. Note that when I say foam, I don’t mean the stuff that keeps the shipment of vintage Superman comic books you just bought on eBay safe. Sound waves just laugh at that stuff as they pass effortlessly through it. If you have outfitted your studio walls with egg cartons, take 30 seconds to hang your head in shame, and then go immediately and take it down. Go ahead, I’ll wait…

My point is that if you use foam, there is simply no substitute for the stuff that is designed specifically for sound treatment. If you want to go the DIY route, then rigid fiberglass is your friend.

Excessive ambience: By this I mean general undesirable reverberation from the sound bouncing around the room in a broad range of frequencies. Unless your room is tiled, or has very dense wood paneling, this is probably not your main problem, but even a small room may have excessive amounts of bad reverberation – the kind that makes what you record sound boxy and indistinct.

A bit of absorption and/or diffusion on the walls is a cure for this. Again, homemade absorbers and/or diffusors as described above will probably take care of it, or if you like, some commercial studio foam.

Comb filtering: a hollow sound caused by some frequencies being canceled and others being emphasized as the sound bounces around a room. If you graph the frequency response, it shows closely spaced peaks and valleys, and looks like a comb.


Frequency buildup and cancellation in a comb-filtered pattern

Comb filtering was the bane of my audio life for years.

By necessity, I track vocals and acoustic instruments in an isolation booth. (I live on the flight path for a large airport, I’m only about 500 yards from a railroad line, there’s a busy road nearby, and I have neighbors who like to use lawnmowers and other loud tools at the most inconvenient times.) My booth does an excellent job of keeping all those external noises out of my recordings.

Unfortunately, it also has a very boxy, unpleasant acoustic signature, caused to a great extent by the comb filtering that results from non-random reflections causing narrow bands of frequency cancellation and emphasis. It’s impossible to fully deal with the frequency imbalance with EQ, because there are narrow peaks and valleys all up and down the spectrum. I have spent many happy hours playing “whack-a-mole” with comb filtered vocals recorded in that booth. As soon as I thought I had dealt with one problem frequency, another would pop up.

I have to confess that I never found a completely satisfactory DIY solution to this. Getting the room as dead as possible seemed to be the most effective solution for me. In a typical booth, diffusors are problematic, because effective ones eat up too much space. Finally, I ended up purchasing a set of eight stand-mounted tube traps, that flood the area with random reflections that completely eliminate the comb filtering. This is not a cheap solution by any means, but it really does work.

Room modes: based on the geometry of the room, it may resonate at certain frequencies. The smaller the room, the higher the frequencies tend to be, and therefore more apparent. Room modes are fairly predictable based on the dimensions of a given room. This site and this site both show how to calculate the modes for a room. Once you know what frequencies your room wants to reunite at, you can build absorbers and diffusors that cover that range.

Keep in mind that the above is a simplification. But if you take some basic steps to treat your recording space scientifically – i.e. not by gluing random stuff to your walls – you will certainly improve the quality of your recordings a lot.

Endnote:
Here are some links to information from people who know way more about this than I do:


Jeff Bohnhoff is a musician, audio engineer, and record producer from California. His and Maya Bohnhoff’s latest CD is Grated Hits; their albums can be purchased online at CD Baby. Follow them on Twitter or on Facebook.

 


This post is part of The DIY Studio Buildout Series, on building out a home recording studio.

studio buildout part 6: your computer and digital audio workstation

We’re heading up to Vancouver tomorrow for VCON! We’ll be there for the weekend, hitting Chapters and Siegel’s Bagels and picking up some desperately-overdue cider rations and kicking around town. Mmm, Growers, how I miss thee. If you’re around, yell!

Also, there’s an exciting special event coming up here next week; you’ll want to read about it. More on that below the fold.

Right now, let’s talk Digital Audio Workstations.

First, what are they? Simply put, Digital Audio Workstations are software implementations of the physical hardware you’d use in a large recording studio to record your music. They include virtual mixing board, virtual patchboard, virtual tape recorder, virtual cables, virtual effects plug-ins, virtual equalisation – and depending on the package, even more.

The goal is simple. If you can do it on one of these:


I’ll be in my bunk

…then you should be able to do it in your digital audio workstation (or DAW) software.

Of course, it’s not quite as simple as basic recording. Were it, you could get a little digital recorder and be done. What that giant hunk of hardware – or your software DAW – gives you is the ability to record several tracks of sound, separately or all at once.

A DAW lets you play those tracks mixed together in a synchronised fashion, move and edit your recorded sounds, adjust their levels (both relative to each other and in absolute terms), adjust equalisation, add effects such as reverb or distortion or overdrive or whatever you have plugins for, and so on.

Some DAWs include integrated MIDI support; some include sequencers as a core component. Some even support remote boards that give you all those sliders and knobs, so you don’t have to use the mouse or keyboard so much. Those are cool, and easier to use in some important ways, if less portable.

But at the most basic level, you have recording, editing, mixing, and playback. At the most basic level, you have GarageBand.


I will not be in my bunk.

Now, I’m not mocking GarageBand. GarageBand is a great introduction to concept, and surprisingly capable. It makes a whole bunch of tasks really easy, has integrated MIDI support, and includes a bunch of virtual MIDI instruments.

While from a features standpoint it’s pretty limited, and while it handles tracks in a way that implies they’re less generic than they are by naming them after instruments and making them sticky in weird ways which might confuse you later, it’s still a great first experience.

If you just want to get the idea with GarageBand before tackling something more complex? Go right ahead. Because I am not going to lie to you: the learning curve on the more advanced DAWs can be brutal. Particularly on the free/open source ones.

So, what’s out there? Well, if you have the money, and a Mac, I hear great things about Logic Pro. For both Mac and PC you have Pro Tools, which is called an industry standard because it is one. Pro Tools Express is free with some hardware purchases – but it’s also limited enough that I wouldn’t use it myself. Reaper, for Mac and Windows, has fans in the professional community. (And as Tom Smith noted last week, IK Software is having a big sale right now. This is relevant to your interests.)

But we’re about dirtball DIY. Let’s talk building your own kit, and doing it the cheapest way.

There are really two topics here: hardware and software. We’re already talking software, so let’s carry on.

The cheapest route, in dollar terms, is always open source. Linux is free software. You may have to be able to do a lot of internals work – no, that’s not fair; you’d better be ready to rip its guts out – but you can do it.


Afraid? You will be. You will. be.

Audacity is a relatively-simple open-source DAW. It runs on Windows, OS X, Linux, and some Unix OSes, not that you’re likely to run into those. It’s easier to set up and it works. I ran into its limitations in the first hour, but that’s because I already had aggressive goals; it’s the GarageBand of the open source world.

Ardour is my workhorse, and it is a monster. It runs atop specialised sound server software called JACK, and runs on OS X and Linux. If you run it on Linux, you’ll have to grab PulseAudio by the throat, slice off its head, and salt the ground on which it dies. This will not be easy in some Linux variants (Ubuntu, I’m glaring hatefully in your direction) but it must be done. Ardour is monstrously frustrating (at times), is possibly the most difficult to learn software I’ve ever used outside of 3D modelling…

…and it can do anything. But it will make you cry getting there.

MusE has a fair bit of traction in electronica, because it’s really a sequencer. But it also has DAW capabilities, and the stated intent is to expand into the DAW arena. It’s Linux-only. If you anticipate a lot of sequencer use, and have relatively light physical instrument requirements, give it a look.

Rosegarden started out as MIDI and composition software, and that’s still where its heart is. But, as with MusE, it’s headed into DAW territory and added at least some of the basics of the functionality. If you like sheet music composition and MIDI, you may want Rosegarden.

So, what about the hardware? I’ll approach this from the idea that you’re building a new box for this, or upgrading an old one substantially. If you’re not, well, skim this anyway.


Screw you, Best Buy

Here are things not to care about: what the case looks like. How cool anything on the motherboard sounds. (We already talked about external sound interfaces; if you skipped it, go read up.) The graphics card. You’re not doing video: you do not care.

What you do care about: fan noise. Bus throughput, on the hard drive side and on the USB chain side. (I’m assuming you’re on USB and not FireWire or Thunderbolt, mostly because that’s where we are in the technology curve right now.) Raw CPU power. Lots and lots of RAM. If you want to spend some money, throwing some dosh at an SSD drive is not misallocated funds.

Basically, you want to build a lean box dedicated to math – because math drives your virtual effects – and moving audio data around, and nothing else. Every other toy, every other frob, adds interrupts and takes CPU and bus time away from what you’re doing with audio. Rip that shit out.

One particular task you’ll want to figure out is probing your USB bus for onboard devices. A lot of motherboards will share device assignments between on-motherboard equipment and external USB ports. This is technically correct – the best kind of correct – but in high-demand applications results in more interrupts on the bus and slower throughput. This can and in my case did result in higher latency and buffer overruns. Find and use ports which are unshared for your external audio card.

Also, for Linux in particular, you may find that wireless internet will be a problem. It’ll work, but will interoperate badly with your realtime kernel, hammering you with interrupts and popping you out of realtime mode.

Some people ditch networking entirely. If that’s not okay, go wired. If you must go wireless, get an external wireless bridge and connect it via ethernet cable to your wired (and realtime-kernel-compliant) ethernet card. This will solve many weird network problems.

But I said we’d talk about hardware, dammit! So okay! Where do you get performance hardware for cheap?

Well, you shop around, of course. Check your local parts stores, but the cheapest route I’ve found is to get a copy of CPU magazine’s motherboard roundup issue – preferably the last couple of years’ worth – and to go the gaming kit-out sites.

Yes, I know, I just talked about case mods and all that: don’t care. You don’t go for the frills: you go there for the motherboard clearance sales, because last year’s gaming l33tness is this year’s dogshit, as far as they’re concerned, and they just want it gone.

As a result – the fire-breathing motherboard inside my DAW? 75% off retail. The CPU, 60% off. The RAM, sadly, not as much, but still: bargains are to be had, and I had them.

When browsing, though, choose wisely! Look over the supported hardware list for your operating system and DAW and follow them. The last thing you want to be doing is tracking down some obscure kernel bug and finding that it’s only fixed in a downstream revision your distribution doesn’t even support yet, so you end up installing a custom kernel configuration and doing haxx0r insanity, not that I know anything about that.


Fuck yeah, meme baby. Fuck yeah.

And that’s an overview! Believe it or not, that is an overview; there are an endless series of twisty passages you can run down on this topic, all alike. I’d browse a little, pick one, and dive in.

If you’ve already built a DAW, what do you use, and why? What problems did you hit that I haven’t covered? Is anybody out there using Thunderbolt yet? Share your experiences!

Finally, I teased an announcement up top. It’s super awesome. Get this:

NEXT WEEK, we have a special event! We’ll be kicking off a series of monthly guest DIY posts with one from JEFF BOHNHOFF.

You may know Jeff and Maya Bohnhoff from their YouTube hit, Midichlorian Rhapsody, or some of their many albums and awards. Jeff and Maya also built Mystic Fig Studios, and Jeff has engineered and recorded literally dozens of albums in his 30-year musical career.

And next week, Jeff will be stopping by here, to talk about DIY sound control in your home studio. We’re thrilled to have him, and YOU WILL WANT TO READ THIS, if you have any DIY recording interest at all.

Until then – see you in Vancouver!

 


This post is part of The DIY Studio Buildout Series, on building out a home recording studio.

studio buildout, part 5: sound interfaces

Hello, Thursday! Yes, I know, DIY day is Wednesday, I was busy, with Stuff. I DIDN’T FORGET YOU GUYS! <3

Last week, we talked about microphones! As part of setup for that, we talked about XLR interfaces and balanced signals. If you missed it, go read up on that.

Now, let’s talk about why you really, really want an external sound device, rather than using your super-l33t gaming sound card. I mean, you paid good money for that thing, right?

Well, aside from the connectors and signal types, there’s noise. The inside of a computer cabinet is really, really noisy, from an electrical standpoint. And microphone signals are really, really small. The balanced signal noise cancellation falls over as soon as you hit the connector, so you don’t have that protecting you. And if you’re recording, the last thing you want is unintentional electrical noise on every track.

Having the sound card be outside the box, and converting everything to digital before it gets to your computer solves all those problems. It also lets you have the computer in a closet, where its fan noise and hard drive noise are nice and safely locked away from your microphones, and where it is safely out of the way your crazy bassist who likes to kick things.


Also, from this guy.

But aside from that, let’s talk goals again. We discussed goals quite a bit in monitors and monitor amps: sound equipment is built to particular goals. Onboard soundcards are built to make cheap computer speakers sound better; gamer kit cards are built to make games sound awesome. And those are both really good goals!

But they are not your goals in the studio. You need sound equipment that is precise, and which treats different sounds similarly, across the frequency spectrum. You need AtoD and DtoA converters which are “musical,” which is to say, are accurate and even-tempered. You don’t want help, because you can’t be assured of getting it out in the wild when people are playing your music back.

I mean, if it’s all you have and you really, really can’t afford anything else? Fine. Of course you should use what you have. Chiptunes people can do this pretty effectively, as can anybody not using microphones or live instruments. Use what control you have over whatever sound card you have to minimise or disable all “sound enhancement,” “bass boost,” “loudness,” “surround effect,” anything like that you can find. Turn all that shit off, dig until you’re sure there’s nothing left to turn off, and you can do okay.


Honestly, people, it’s not that hard

But let’s say you’re not doing that. What do you need in one of these?

First: at least two inputs. If you’re willing to stick to recording one person at a time and not recording a drum kit, you can get away with only two. The inputs need to support XLR connectors. Almost all these days will also support TRS (a.k.a. 1/4″ plug, a.k.a. “patch cord”) connections on the same inputs; it’s a combined socket and really clever.

“But Solarbird!” I hear you cry. “You just told us last week, never use TRS connectors!” Wrong, minion! I said, no microphone worth the time will have those, and that’s still true. But a lot of other devices will have them – synth, electronic keyboards of other types, drum machines which aren’t purely software, Weird Shit You Build Yourself – it’s a big list.


Optical theremins do need apply

You can even connect an electric guitar straight to one of these, and people will do that. If you’re a classic rock guitarist and want to sound like Tom Scholz? Now you know. (Okay, it’s a little more complicated than that. BUT NOT MUCH.)

Second: Those input connectors need to support phantom power. Phantom power needs to be switchable (on and off) separately to the device as a whole. And it should be 48v. There are specs now for lower-voltage phantom power, but a lot of equipment won’t work on it.


I see what you did there

Phantom power is a way of throwing DC power on the line in such a way that it’s invisible to the audio signal, but can still be used by the condenser microphone connected to it to power the condenser pickup. We talked a little about this last time.

Third: A headphone jack which includes passive or live or real-time monitoring in the unit itself. This requires some explanation.

When you’re doing multitrack recording – which is to say, recording one instrument, then another instrument, then vocals, all separately – you need to be able to hear what you’ve recorded so far, on headphones. These headphones need to be pretty sound-tight so that what you’re listening to doesn’t get picked up by the microphone again and re-recorded.

But you also want to be able to hear yourself, and good headphones will block a lot of the sound you’re trying to record as well. Trust me, this is important. You further need to hear the sounds you’re playing as you’re making them – like in real life – without any processing lag.

If your sound interface sends the microphone input to the computer and has the computer send it back for monitoring purposes, that will take enough time and introduce enough lag that it will really screw you up. Seriously, it’s like a tenth of a second or more.

So a decent external sound interface will provide the ability to throw both playback and what’s coming in the microphones back into the headset at the same time. Playback and microphone monitor levels should be separately controllable, too.


You’d think somebody would have a picture of a monitor lizard wearing headphones, would you. WELL, WOULDN’T YOU?! All I could find was this rather nicely-rendered Gecko.

Fourth: Studio monitor outputs. These will usually be RCA connectors, but might be TRS connectors. They’re for playing back things you’ve recorded on an amplifier – your studio monitor amp and the speakers connected to it.

Finally: Good quality analogue-to-digital converters and digital cable connection to your computer. You can get away with USB 1.1 equipment if you’re down at two channels or fewer; more than that, USB 2.0 is a bare minimum. Firewire and Thunderbolt are of course both better, but unless you’re working on a larger scale than anybody I imagine reading this will be doing, unnecessary.

Those are the required features. There are other options nice to have; inserts (to add effects boxes live on your inputs, if you feel the need to do that for some reason after input), pad controls for particularly “hot” input, things like that. They’re nice, but less important.

Plus, of course, more input connectors! Why that’s better should be obvious. But I need to point out here that more is not intrinsically better. Every additional input requires duplication of an entire channel of circuitry. Remember back on monitor amps, where I showed how the left and right amplification channels had mirrored circuits? Each additional input has another set of input channels, just like that.


More != Better

Those cost money. At any given price point, there’s only so much money to spend on hardware. So if you have two inputs on a $250 device, you have a lot more hardware money to spend on the quality and feature set of each channel than you do if you have, say, eight channels on a $250 device.

Particularly at our budget, small differences in money make big differences in quality. Let’s take a couple of examples; I have a TASCAM US-800 (eight channel, six with microphone preamps) and an M-Audio USB Fast Track Pro (two channel, effectively).

The US-800 listed new for $370, was discontinued about a year ago, but is still floating around on clearance new for around $200. The Fast Track Pro retailed – I believe – for $280 originally; it’s floating around for $150-$200-ish. Both have all the basic features listed above. The US-800 is USB 2.0; the Fast Track Pro is USB 1.1.

If you do the math against retail – which is our best ratio, for getting at manufacturing cost – you’re spending $46.25/channel on the US-800, and $140/channel on the Fast Track Pro. And that shows up. Some of it is in features per channel; the Fast Track Pro is quite feature-rich for its price and size.

But it’s also audible. You hear it in the quality of the microphone preamps inside.

Don’t get me wrong; the US-800 is good. At lower gain, it’s very good. I use it heavily. It’s fast, it was hell and a half to get working on Linux (hi, I have a custom kernel configuration now!) but it works. But despite being more expensive overall… it’s just plain noisier, at high gain. You simply can’t boost the microphones as much you can as on the Fast Track Pro.

So if I need extra mic gain, and I don’t need more than two inputs, I’ll hop over to the Fast Track. It has fewer inputs and is stuck at USB 1.1, but also preamps that don’t add noise at high gain. As always, it’s a matter of making the right tradeoffs, and picking the right tool for the right job.


To wit

So that’s a basic overview of audio interfaces! We had some great commentary last week on microphones, mostly on the Livejournal echo, but also on Dreamwidth, including ideas for making your own pressure-zone microphone out of piezoelectrics and glass, a lot of commentary on micing bodhran, some thoughts on pickups, and the sudden and strange return to popularity of the ribbon microphone. If you wanted more on microphones, go check out those discussions!

Next week, we’ll talk a little about cheap/open source digital audio workstation software. And, of course, if you have any thoughts or questions on sound interfaces, let’s hear ’em! Some of you guys are recording, what do you use?


Added May 2013: The original version of this article mentioned crosstalk in the US-800. This turned out to be an unrelated wiring problem, and not intrinsic to the TASCAM unit itself.
Added January 2014: The noise issue at high gain got substantially quieter with the addition of ferrite chokes on all power cords. Turns out the building’s wiring is genuinely rotten with RF! A better board wouldn’t’ve cared, but this one does. Still, it’s a cheap fix – $10 for 10 chokes at Amazon.

 


This post is part of The DIY Studio Buildout Series, on building out a home recording studio.

the future is very fucking nigh

I was going to post this week about microphones, but I’ve been fighting some sort of nasty head cold all week, and really just have no brain for writing right now. So, next week in DIY: microphones!

Fortunately, I have something easy and short to write about.

Last week I had a guest post from a spambot. Okay, no, it wasn’t a guest post from a spambot. It was a spambot’s comment, however, that was not merely on topic, but useful, relevant, interesting, and started a valid argument.

Instead of approving it, I elevated it to top level (spam content graphically blocked out) and said “your move.”

I should’ve kept my mouth shut.

In response to this post on acoustic sound dampening for your DIY home studio, the same spambot – and from the blanked out material, it’s clearly the same bot – has this to contribute:

Text:

I recently finished a 10 x 18 room, with some guidance from ReadyTraps (they will help with cad design and useful advice for small dollars, nice products too). In a nutshell, I doubled the drywall with “Green Glue” in-between, added about 12 2×4 by 2″, 6 2×4 by 4″ panels of 703, some on the wall, some spaced. Then two wall to ceiling superchunks. Is it acoustically perfect? NO, but it sure is predictable for monitoring, and dry enough to get very clean live tracks that do not have the boxy home studio sound. My room still booms in the sub 100 range, but its not hard to mix around. In my mind, a workable room that can be had for a couple grand, in combination with the great DAW’s available now, is what is bringing recording to the masses. Yeah, we all get to play now! I think the article would be more aptly named “Why your bass traps don’t work perfectly”.

I want to note for the record that the spambot’s actual spam content had nothing to do with Ready Traps or GreenGlue.

Really, I don’t have much of anything bad to say about this at all. Corning 703 is a good rigid sound-absorber, and you’ll find plenty of DIY centred around using it for sound panels. It’s good material, and not very expensive for those 2″ sheets they’re talking about. More expensive than my carpet baffles, but you’re not gonna break the bank.

The only thing I’m not sure about at this point is how a spambot does physical construction, but then again, I’m making assumptions. Maybe this is in the virtual world. In which case, hey, Smartbot! Say hi to Tron and Ram for me, would you? It’s been a while. ^_^

i did not come from a human mother
i am the speed, the information i collect
and i can do anything i want

gonna be the future soon

Remember this XKCD?

I am staring at Mission. Fucking. Accomplished. in my spam queue right now. In response to my post last week about studio monitors and the importance of flat frequency response curves:


The text:

Why have I included a frequency-response curve here? I mentioned earlier that the frequency-response curves in a sales brochure are typically meaningless in terms of providing information that’s useful to an end user. Actually, though, I’d go further than that, and suggest that in many respects making any judgment about the worth or likely value of a monitor by examining its frequency-response curve is not far short of pointless. I often read opinions on the SOS Forum arguing that to be of any value monitors require a ‘flat frequency response’, but numerous recordings made during what many would consider the golden age for musical sound quality (the ’60s and ’70s) were monitored on speakers that were all over the place in terms of frequency response — and I don’t know why recording engineers seem to believe so strongly that a monitor should be anechoically ‘flat’ when so much end-product evidence suggests that this isn’t particularly important.

Constructive. Relevant. Interesting. Starts an argument. And the blocked-out information reveals it to be a spambot.

I kind of want to approve it! I’m not. I’m going one further: elevation to top of post, and addressing the spambot’s point, since it had one. Congratulations, spambot, well done: you’ve earned it.

My response, I suppose, would be that the nonflat monitor speakers of the time were reasonably accurate representations of average home speakers, which were nowhere near flat either.

And once you got into the era of flat response curves being achieved, followed by an era of goosing-by-design (rather than nonflat-by-technological limitations), it became necessary to move to a neutral reference base for studios. Simply put, you can’t try to guess all the many ways that people intentionally-off-flat-response systems, so don’t try to optimise for any of them; optimise instead for the average of all of those systems.

I’d also argue that the late 60s weren’t my idea of a golden era of recording. There are some fantastic jazz and classical recordings from the era, absolutely, but a lot of rock and pop was still very fuzzy and often kind of muddled. To my ear, recording continued to improve up until the Loudness Wars – with a hiccough as everyone learned to deal with digital equipment – and that’s fashion, not technology.

So that’s why I still argue that in the current era going for flat – or reasonably close to it – is the best idea.

Your move, spambot. I’ll be checking the queue.

it’s not your fault… or is it?

studio buildout, part 3: playback amps

Woo! We’re Newfoundland and Labrador Folk Festival official photos page. Wait for it and you’ll see Anna and me both! Front page! 😀 We’re in fact in several photos here – one where I’m performing solo (my extended version of Ten Finger Johnny) and later with Anna in session. 😀

(Sadly, the photos – by Rick West – were taken down in when they redid their website)

Newfoundland music on Newfoundland soil. That’s called the correct way Even if I do have an uncanny ability to blink just in time for the photo. XD

But now, back to business.


Last time, we talked about monitor speakers; what to look for when you have no money, characteristics to seek out, simple mods to improve their behaviour, and so on.

But unless you went with powered monitors, you’re going to need amplifiers to drive those monitor speakers. Since you’re reading this, you probably aren’t going to just go out and pay full retail for some very nice new equipment; let’s talk DIY!

First, I need to repeat something I said last time:

The cheap but rebuildable equipment you want mostly comes from the 1970s… There are a couple of key reasons for this: 1. By this time, transistor audio technology had settled down, and no longer sounded like ass. 2. The state of the art was finally good enough (in transistors) that the then-goal of broad and equal frequency handling – meaning, flat audio reproduction curves – became realistically attainable, and people were still trying really hard for it.

This is true in amplifiers, too. Some would argue that in amps, you want to stick to the early 70s. I don’t particularly agree, but be careful when you get into the early 80s, just because of audio fashion trends being what they were.

You can also step back a bit into the 1960s, if you’re willing to learn vacuum-tube equipment. In some ways, that’s easier to work on, and you’ll get fantastic bang-per-buck. Look for EICO, Dynaco, Harmon-Kardon, just for examples; and research tubes first, to see what’s back in production.

Tube equipment has downsides, though: you can’t tip them on the side, they use a lot more electricity, need more ventilation space, generate a lot more heat, and most importantly of all, the power rail tends to be hanging out in the 450 volt range. Careful with those pliers!


Think of it as the advanced class

So unless you’re okay with that, stick to the transistor era.

If you poke around, you can find a pretty good number of old 70s component-stereo-system amplifiers for very little money. Don’t buy the combined units, with turntables and tape decks built in; those were junk then, and are junk now. You’ll see nostalgia for some of that era, and entertaining tho’ that might be, it’s not our goal. Look for something that’s just amplifier and pre-amp – preferably a unit without even a radio.


Undeniably groovy, but still kinda terrible

Pioneer is usually a good, safe bet, as brands of the era go; it’s right in that sweet spot of quality and commonality. So is Harmon Kardon. Sansui, Kenwood, and Marantz are often excellent, but tend to cost more even now. My general approach is to keep an eye open and when I see something of the right sort, then search the web for it and see what people have to say. AudioKarma and Gearsluts are both pretty good data sources in this regard.

My current studio monitor amp is a Pioneer SA-5200. It was made for all of three years (1972-1975) and I picked it up at a thrift shop for all of $5. They go for under $35 on eBay, working to various degrees.


Not mine, but same model. Not so groovy, but far more competent.

It has no power to speak of (20w), but you don’t need it for this application; most importantly, it’s noted for being a very clean amp; very low distortion and very low noise, at least as it shipped from the factory. And it has enough power to drive all my mains, and reference headphones.

That said, it sounded pretty terrible when I bought it, and got worse over time. This is where you need to know something which may and may not make any sense to you, depending upon how much you know about electronics: electrolytic capacitors age and die. And every audio chain you’ll find in any of these amps uses lots of them.

You’ll have to rip out and replace every one.

I’ve talked often about how the most important item in your studio toolkit is the soldering iron. Amps of these vintage can be rebuilt, without complex tools. The parts are large and relatively easy to access. You’ll want a low-wattage soldering iron, so you don’t damage the board with too much heat. You’ll want direct-value replacement swaps on those capacitors, in terms of uF rating. (You can go higher in voltage if you want; that’s a matter of how much the capacitor will tolerate, so replacing with higher voltage is safe.)

The electrolytic capacitors look like this, on the circuit board:


Or Doctor Who. Are you The Doctor? No? Don’t reverse polarity.

Coming out of the bottom of each of those cylinders are a pair of metal wires. Those go through the circuit board and are soldered into place, making contact with the printed circuit on the other side of that board. You’ll need to de-solder those connections, pull up the capacitor, and replace it with caps of the same capacity.

As a side note, these are not the only kinds of capacitors. You’ll see many flat discs; those are ceramic capacitors. Barring physical damage, you’ll never need to replace one. Similarly, you’ll occasionally find flattish rectangular capacitors. Those are usually film, and again, leave them alone, they’re fine.

Doing all this is kind of a pain in the ass, but you generally need to do it in equipment of this vintage. Here’s a bit of a map:


It’s dangerous to go to Toshi Station alone! Take this.

Any stereo amplifier is really two amplifiers combined together into a single box, one for the left channel, one for the right channel. You can see above how this results in symmetrical layout of components! Anywhere you have that kind of symmetry, you’re dealing with the left and right channels, duplicated. Anywhere you’re not seeing symmetry, you’re probably looking at power circuits.

Advanced students will want to bypass the tone controls. There’s no single way to do that, so I’m not going to post pictures. But I will explain why: it’s because, as with the monitor speakers, you don’t want help. You want flat response, or as close as you can get to it. The ideal studio monitor amplifier would be a wire, with gain – that is, a wire that magically changed nothing about your sound other than volume.

Tone adjustment knobs and systems, by definition, deviate from flatness. They’ll also add noise, so just bypass them. It’s also one less set of components to rebuild, so saves you time!

And that’s how to get a quality monitor amplifier on the smallest budget – at least, that I’ve found so far. Next week: I dunno! Microphones, or possibly digital audio workstation software and computers to run it on. One of those. Happy rewiring! ^_^


ps: Let the kitty help!


No, no, not wire snips – can opener! Here, I’ll get it.

 


This post is part of The DIY Studio Buildout Series, on building out a home recording studio.

studio buildout part 2: monitors

Building out home studios has become de rigueur for musicians of all kinds of levels. This is part two on a series of doing it on the really cheap.

Last week, we talked about the room itself. That’s important, so if you missed it, start there.

But this week, let’s talk monitoring. You already know you need microphones and a sound interface and some sort of recording kit (In free software, I suggest Ardour, if you can make it past the learning curve), but hearing what you’ve recorded being played back is just as important.

Despite this, a lot of people – including me – will try to work off studio reference headphones. Don’t get me wrong, you’ll need those, particularly for listening to tracks you’ve already recorded while playing out the next track you’re adding. Shure makes a nice pair, the SRH-440s, occasionally discounted as low as $50ish.


No joke here; just decent basic headset

But you’ll also want speakers. The audio experience is simply different, and it’s different in important ways. Ideally – particularly if you can’t afford a mastering pass but want to come as close as you can – you’ll have a bunch of different kinds, from crap laptop and computer desktop speakers (critical, given how much people listen on those horrible things) up to some genuinely good pairs of different quality levels.

But this can be a many-thousands-of-dollars project! If you can’t spend any money, what do you do?

The easiest and arguably best thing to do, if you have some money, is to research and buy a good set of self-powered studio monitors. These are speakers with built-in amps, and they’ve become rather the standard. The amps can be tailored to the speakers, which can in turn be tailored to the cabinet in which it’s all mounted. It’s your plug-and-play solution. Hie thee off to a good equipment seller and have at.

But if you’re reading this, you’re probably more of a hax0r, and want to DIY it. Or you just have to, because you have no money to speak of.


Or possibly YKINMK, but that’s okay.

Okay, first, let’s start with an overall tip: the cheap but rebuildable equipment you want mostly comes from the 1970s and early 1980s. There are a couple of key reasons for this: 1. By this time, transistor audio technology had settled down, and no longer sounded like ass. 2. The state of the art was finally good enough (in transistors) that the then-goal of broad and equal frequency handling – meaning, flat audio reproduction curves – became realistically attainable, and people were still trying really hard for it.

Seriously, “reproduce all frequencies, high and low, the same amount” sounds obvious? But that was difficult. People would even print their equipment response curves on packaging.


what you want


what you don’t want

And this last bit is really important, because once reproduction technology really got under control, manufacturers started realising that they could make their systems sound better by not having flat response curves across all frequencies of sound. They’d intentionally boost attractive frequencies a bit, nudge characteristics around – all to make the system sound better.

And that’s great, unless you’re in a recording studio, where you really want that flat response curve. If you sound good on that, you’ll sound good all kinds of places. When you’re recording, you want accuracy, not help.

So. If you have virtually no money and want to do the best you can with a single pair of speakers, look for a pair of these little beauties on eBay:


hey kids, did you know radio shack used to make radios?

These are the Realistic Minimus 7, introduced in 1978. These particular speakers look like they’re from the 80s; the originals had wood cases, not metal. You’ll also see white metal cases, instead of black, and they tend to be cheaper for no functional reason.

And they are the best audio devices Radio Shack ever made. Seriously, when introduced, they showed them off in stores by discreetly placing them atop a pair of massive fuckoff four-way monster speakers on one of their best kits, blasting, with a sign saying ALL THE SOUND COMES FROM THESE —> pointing at the tiny speakers.

This isn’t to say they’re perfect. Far from it. Sound response drops way off after 70-90 hz and there’s nothing to speak of under 50. But in new condition, they are probably the most precise speakers you will ever hear for under $500.

So get a pair of these. You’ll spend $20 if you look around enough.

But there’s a catch, of course. Notice I said in new condition above. These won’t be.


just walk away

The original crossover design in the Minimus 7 is ultra-minimalist (hence the name), which is part of the brilliance of their design. It also used something called a nonpolarised electrolytic capacitor as part of the sound circuit. These components age, and age badly.

So once you have your Minimus 7s, start googling around for “minimus 7 crossover upgrade kit.” One kind will be a direct replacement of the capacitor with a new film capacitor pair; these sound awesome, are really cheap, and leave the speaker with its original response curve. But it won’t be completely flat. Another will be a more complex and expensive kit, which will include a coil; it’s usually called something like a Zobel Network Crossover. That will get you your flattest curve.

And you get to make a decision here which way you want to go. Either way has its advantages, and the decision’s up to you.

If you have a little more money, do the direct-replacement upgrade on the 7s, and then also look for a pair of Minimus 11 or Optimus PRO-x77 speakers. The 11 was a larger version of the 7; it has a bit better bass range. The 77x was an attempt to merge the two product lines; it was not very successful, but can be salvaged.


takes some work, tho’

You can find both of these, too, for around $20, but be careful with the x77; the foam on the woofers can degrade over time. (Which is what that picture above was about.) The 11 didn’t have this problem.

For either the 11 or the x77, however, get Zobel-type crossover replacement kit. Both of these have better low end response than the 7 series, so you’ll get a wider area of flat response curve. A completely upgraded PRO-x77 (with the degrading foam replaced, in particular) or 11 make lovely, lovely studio monitors.

Then grab a pair of cheap computer speakers from RePC or free from Craigslist. It doesn’t matter what you get, as long as they work as originally designed. You won’t mix for or on these, but you’ll test against them occasionally to make sure what you’re doing can still be heard.


yes, yes, too easy

As is probably unsurprising, I have a setup kind of like this. I have a pair of AFCOs rather than Minimus 7 speakers, but they’re similar devices, and have been similarly modded. I have a pair of Optimus PRO-x77s that needed new woofers but now have nicely flat response curves; they’re becoming workhorses.

I also have a pair of Bose 301 that I mostly use for checking out bass, because even modded, the x77s aren’t awesome on the low end. (But the Bose are not good monitor speakers overall – they “help” – so I don’t rely on them.) And finally, I have a pair of junk powered computer speakers that I fixed and modded to interface with my studio monitor amp while still using their built-in amplifier.

I got the AFCOs for free; no, wait, it was better than that. I found them, abandoned, outdoors, in Seattle’s U. District. The mod kits were $12, and now they’re two of my favourite listening speakers. The computer speakers, I’ve had since my Amiga. The x77s – I don’t even know. I’ve had them sitting around in a box as spare/junk speakers for just forever, not even realising that with a few mods and a few hours work, they’d suddenly turn awesome.

So now I have a four-sound-profile mini-mastering setup, all for on the order of $150 out of pocket, including speaker wire. Is it BEST SETUP EVAR?! Hell no. But it’s genuinely pretty good, gives me a variety of listening models, and I sure wish I’d had it while recording Dick Tracy Must Die. You’ll hear the difference that a better setup can bring on Din of Thieves.

Next time: if you didn’t go with self-powered speakers, you’ll need a monitor amp! As always, I have some suggestions. Ja ne!
 


This post is part of The DIY Studio Buildout Series, on building out a home recording studio.

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